similar to: Meetme and MEETME_EXIT_CONTEXT

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "ua1" calls user "ca1" 2. "ua1" then presses the feature code "*0" to redirect "ca1" to conference room 300 3. "ua1" then dials the user "33" 4. user

Hi, i am using the n-way-call dialplan solution found on voip-info. i have added its entry in applicationmap of features.conf file. the problem is......its not working. to activate the n-way call i dial *0 but nothing happens. i have played around with dtmf and codec settings but no success. the extensions and sip configuration is below if you want to have a look. I dont have any clue why its not

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "user1" calls user "user2" 2. "user1" then presses the feature code "*0" to redirect "user2" to conference room 300 3. "user1" then dials the user "user3" 4.

Hello! I need to use n-way call as it described here: http://habrahabr.ru/sandbox/52259/ It is in russian, but dial plan is quite clear. It works in asterisk 11: -- Blind transferring OOH323/7272-6385 to '0' (context fromtransfer) priority 1 -- Executing [0 at fromtransfer:1] NoOp("OOH323/7272-6385", "") in new stack -- Executing [0 at fromtransfer:1]

Hi All, I download the app_asyncgoto.c, compile the app_asyncgoto.so. Then according to this page http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ; when I dial ,there have this warning: -- Executing AsyncGoto("SIP/111-086497c8", "SIP/113-08674628|dynamic-nway|111|1") in new stack Feb 2 16:53:10 DEBUG[4218]: app_asyncgoto.c:95 asyncgoto_exec: Attempting

running Asterisk 1.8.9.0-rc2, what are the ways to interface with ConfBridge ? I see the CLI command 'confbridge' documented for asterisk 10, but i dont see how to interface with confbridge on 1.8 What I'm trying to do is keep track of conferences that are used. I tried something like the below, but not only does Confbridge not return, but i'd need something that erases the

Hello, I'm trying to implement multi-party calls according to these instructions: http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO They are almost working, except that the Goto at the end of [dynamic-nway-start] doesn't seem to work. When I turn verbosity up a bit, I get something like this in my error log: == Channel 'SIP/SWG-0085a180' jumping out of macro

I'm using asterisk 1.4.5 , on Centos. My kernel version is 2.6.9-55.ELsmp I've configured the nway call. I made entries in extension.conf, feature.conf, as per required. I'm trying to make a 3-way conference with the 1 user myself ( using asterisk), and two others are PSTN line users. I'm making a first call , then putting that person on hold by pressing **( as per feature.conf

Here in Sweden, it's supposed to be springtime. A wonderful time of the year, with sunny skies and wonderful weather. Almost summer. Today, it's not. It's winter all over again with rain and only 3 degrees celsius outside. Better to stay inside and write a weekly Asterisk newsletter :-) This week's topics: ------------------- * Looking beyond Asterisk 1.0/1.1 - what's up? *

I've just started trying to use this functionality and I get the invalid conference number message. Any ideas? I started out with: exten => 7315,1,Meetme,1234 and confno = 1234 and then tried: exten => 7315,1,Meetme and confno = 1234 and enter 1234 at prompt. All give the same message.

Hi, Can someone set me straight as to how to write formulas in R to indicate: complete independence [A][B][C] joint independence [AB][C] conditional independence [AC][BC] nway interaction [AB][AC][BC] ? For example, if I have 4 factors: hair colour, eye colour, age, sex does > mosaicplot( frequency ~ hair + eye + age + sex) mean that the model fitted is of complete independence of

*** If anyone has a better way of doing this, please post to the list. I hadn't seen anything on this list or in channel.c/chan_local.c - which prompted this email *** I'm not sure how many VoIP providers out there are using Asterisk as a service platform like we do, but I thought I'd share an experience with call looping that was racking my brain with the list. One of the

Hi all, I just posted this question before last week. Meanwhile after upgrading Asterisk 1.0.7-BRIstuffed-0.2.0-RCg to 1.0.8-BRIstuffed-0.2.0-RCh the same problem occurs, but seems to be more seldom. Attached is now the output of "zap show channel" . - I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM equipped with 1x TE410P and 2xJunghanns OctoBRI running in NT-mode.

OK... so I'm not sure what I'm looking at. I've had the good old echo problems on my Rev C FXO again this morning, so I thought I'd attempt some debugging, though I'm not sure what I'm looking at. This call has echo. Channel: 2 File Descriptor: 20 Span: 1I> Extension: Dialing: no Context: incoming Caller ID string: "External Call" <99999999> Destroy:

Dear list, I have a dataset as follow and I would like to count the frequencies for the combination of the two variables (f1 and f2) for each id. I know it is should be straight forward, but I just don't know how to do it in R. Here is the SAS code I will use to get the output I want : proc means nway; class id f1 f2; var flag output out=temp; Dataset: id f1 f2 flag 798 1 2

Folks, I have the following code, that works fine on smaller data sets. For larger datasets, it runs out of memory and runs way too slow because we are essentially creating large vectors with rep() and then calling median() on it. (I learned this approach from a post on the web). Below that, I have written the corresponding SAS code. The SAS code works fast because I can just tell the proc

Hi there, I have some problems when using READ() statement in the dialplan to collect DTMF digits. I'm using the following within my extensions.conf to receive 6 digits exten => 9070,n,Read(CONFNO,conf-getpin,6) So far it works! The user getting the announcement and asterisk waits for 6 digits. User entered "102030", but asterisk gets "102033"!!! Executing [9070

Hi, I'm trying to make a dialup internet connection through my asterisk PBX. When I bipass the Asterisk box, I can get 51600bps. When I run through the asterisk box, I'm limited to about 21600bps. I have a TDM31B card. Any help on speeding these connections up would be good - I was on the understanding that if you bridged the channels, then the call should essentially flow straight

Hi @ all, what is the best way to to use features like MeetmeCount without leaving the conference. I use Meetme(,X) and MEETME_EXIT_CONTEXT=context, but the problem is that the caller leave the Conference :( Is it possible to press a key, without this obstacle? Thanx for your answers Daniel Knoll

In Asterisk 1.2.4 is love being able to recording conferences. However, using the default variables, the files are being written to /var/lib/asterisk/sounds instead of /var/spool/asterisk/meetme. If I change MEETME_RECORDINGFILE variable to something different in works, bit I lose the ability to define CONFNO as part of the file name, which is handy when sorting for users to review. I call meetme